12 Dec
2005
12 Dec
'05
9:49 p.m.
From the log, it showed lookup(): 'account
name' Not found in usrloc.
I think it is the NAT problem, so I use stun for the
case below.
Telephone (A)
|
PSTN
|
G/W
|
openser
|
NAT -- IP phone(B) -- STUN
+ ----- IP phone(C) ------+
Below is the result:
A to B/C is ok
B/C to A is ok
However, there is no sound when B to C or vice versa. What reason
will cause no sound between B and C? Is the the reason from the
NAT/STUN?
On 12/12/05, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
Use ngrep to watch for incoming SIP requests on the
SIP proxy.
Take a look at the logfiles on the gateway.
klaus
unplug wrote:
Below is the common configuration of the
network.
Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger)
I can make call from softphone to Telephone. However, it is failed to
make call from Telephone to softphone. I wonder why it happened and
any reference to trace the problem. Anyone have such experience?
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